[OpenSIPS-Users] Wana replace Asterisk with OpenSIPS in OpenBTS Project

[Bogdan-Andrei Iancu]

Hi,

First of all OpenSIPS is a sip server so it works only with SIP.

Secondly, by default opensips is SIP proxy, so it cannot do handover. 
But using the Back2Back User agent module, you may be able to play with 
the ongoing calls and move them between different termination points.

I can help you more if you could describe the handover scenario you need.

Regards,
Bogdan

ALICOMPUTECH wrote:
> Hello
>       Everyone
>                I want to replace the Asterisk (being used as a SIP Server for registration, authentication and call routing) with OpenSIPS in OpenBTS project, as i am planning to have an Asterisk cluster for dedicated services and OpenSIPS will be forwarding the SIP calls to the cluster.
>
> OpenBTS implements GSM Um air interface and emulate the Mobile handsets as the SIP endpoint and these handsets can be used as SIP extensions in a SIP-capable server.
>
> I need to know the handoff and/or handover support in OpenSIPS as i am a newbie to this wonderful open source solution.
>
> If there is any pointer and/or previously handoff/handover work done please share, it will then ease my work
>
> thanks in advance
>
> Best Regards
>
> Bye
>
>
>
>
>
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>   


-- 
Bogdan-Andrei Iancu
OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and "know-how"